Audio recording system

ABSTRACT

An audio recording system comprises an array of microphones and a processor module. The array is a logarithmic spiral array in which the minimum distance between each microphone and all others that lie closer to the centre is maximised. The processor module further comprises a beam direction control device and a beam width control device; whereby an audio target is isolated.

BACKGROUND

This invention relates to an audio recording system, in particular forrecording a specific audio target some distance away.

There are many situations in the broadcasting industry where it isdifficult, or impossible to place a microphone close to the sources ofsound that are of interest. For example in the coverage of many sportsevents it is impossible to place a microphone close to the players. Inthese circumstances the broadcaster will often use “gun” microphones,which are directional gradient microphones. In the case of footballcoverage, there may be as many as 12 gun microphones deployed around thepitch. These microphones have a number of limitations including the factthat they have relatively modest directivity (especially at lowfrequencies); they need to be physically moved to pick up sound in adifferent direction; they are very sensitive to wind noise; and there isstrong frequency coloration of the output for sources off the main axisof sensitivity arising from variations of their directivity withfrequency.

SUMMARY OF THE INVENTION

In accordance with the present invention an audio recording systemcomprises an array of microphones and a processor module; wherein thearray is a logarithmic spiral array in which the minimum distancebetween each microphone and all others that lie closer to the centre ismaximised; and wherein the processor module further comprises a beamdirection control device and a beam width control device; whereby anaudio target is isolated.

The present invention includes a logarithmic spiral array in which theminimum distance between each microphone and all others that lie closerto a centre of the spiral is maximised, which enables a particular soundsource to be discriminated from surrounding sounds and then extracted,either in real time or by processing stored data. Once a particularsound source has been isolated, the array can be directed to track soundfrom that particular source and exclude other audio sources of lessinterest.

Preferably, the beam direction control device comprises a plurality ofswitched delay elements.

Preferably, the beam width control device comprises beam filters.

Preferably, the processor module further comprises a blocking filter;and an adaptive interference canceller.

Preferably, the processor module further comprises an adaptive blockingfilter for adaptive filtering of an audio target.

An adaptive blocking filter reduces the susceptibility of the system toleakage of the audio target signal into the interference canceller (e.g.as a result of acoustic reflections) or when there are phase errors orvariations in the direction of arrival across the array and sodistinguishes wanted signals from reference signals fed to theinterference canceller.

Preferably, the system further comprises a data store, whereby audiosignals from each microphone in the array are stored for laterprocessing.

This enables multiple signals to be recorded and the ones of interestextracted later, without the need to process multiple signals in realtime.

Preferably, each microphone further comprises an analogue to digitalconverter and a bus interface, whereby digital data is transferred tothe processor module via a bus.

This improves ease of array construction and deployment.

BRIEF DESCRIPTION OF THE DRAWINGS

An example of an audio system in accordance with the present inventionwill now be described with reference to the accompanying drawings inwhich:

FIG. 1 illustrates a block diagram of a basic embodiment of the presentinvention;

FIG. 2 illustrates an example of a logarithmic spiral array for use inthe present invention;

FIG. 3 is a block diagram of a preferred embodiment of the presentinvention; and,

FIG. 4 illustrates in more detail a microphone element for use in theexample of FIG. 2.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT

For the purpose of this application the term logarithmic spiral refersto an arrangement of microphones in a plane in which the density ofmicrophones is greatest near the centre, and becomes progressively lowernear the periphery.

The present invention provides an array microphone which overcomes theproblems of the prior art type and provides a very versatile tool forbroadcast applications. Using audio recording technology, it is possibleto record large numbers of audio channels to hard disk at low cost andwith high fidelity. This allows the post processing of signals from alarge scale array of microphones to isolate the audio of interest.

Microphone arrays have been used in hands-free telephony andteleconferencing and have also been proposed for hearing aidapplications, amongst others. For such applications, the arraymicrophones have relatively few elements, typically 4 to 8, and areconsequently limited in their performance. There have also been academicstudies carried out to develop algorithms and beamforming strategies formicrophone arrays. Much of the work has been based on algorithmsoriginally developed for radar, but with adaptations to deal with themuch wider proportional bandwidths associated with acoustic signals andwith the complexities of acoustic propagation and of the environment(air movement, reverberation etc.).

There also exist multiple-microphone arrays for noise source measurementand characterisation. These are primarily for use by scientists andhence are very expensive and require specialist skills to understand anduse. The outputs of such systems are rarely summed to produce an audibleoutput—they are primarily used for visualising the level of acousticsignal arriving from a particular direction.

None of the research to date has concentrated on the specificrequirements of broadcasters. In fact, the large scale arrays that wouldbe needed to meet these requirements have received little attention inthe research community as they have generally been dismissed as toocostly. However, the cost of high quality microphone elements, analogueto digital conversion and processing has greatly reduced in recent yearsand is likely to continue to do so with the advent ofmicro-electro-mechanical systems (MEMs) microphones and increasingintegration of sensing elements and digital electronics. Furthermore,although the processing could be done in real time, a system capable ofcarrying out timely off-line processing of recorded signals can beimplemented at relatively low cost with current off-the-shelf equipment.

FIG. 1 illustrates the basic components of an audio recording systemaccording to the invention. A microphone array 1 comprises a number ofindividual microphone elements 2 each of which provides an input toindividual steer delay elements 3 which provide direction control 51.One output from each steer delay element 3 is input to individual beamfilters 4 for beam width control 52 and another output from each element3 is input to a single fixed blocking filter 5. An optional store 50 mayalso be provided.

The blocking filter 5 provides an input to an interference canceller 6.A summer 7 combines a signal from the interference canceller 6 withoutputs from each beam filter 4 to produce an audio feed 8 and the audiofeed is also returned to the interference canceller 6 to providefeedback to improve the interference cancellation. The steer delays,filters and cancellers together form a processor module.

Beamforming in a number of contexts (primarily radar) has long beenposed in terms of a generalised filtering problem with coefficients inboth the time/frequency and spatial domains. Furthermore, variations onthe generalised sidelobe canceller have been presented that incorporatefrequency dependence in the constraint (for example in Hoshuyama andSugiyama, Robust Adaptive Beamforming, 2001). Constant directivitybeamformers have also been proposed for non-adaptive arrays (Ward et al,2001—“Constant Directivity Beamforming” In publication “MicrophoneArrays”, Springer-Verlag, 2001, ISBN 3-540-41953-5). The presentinvention provides a microphone which combines the concept of ageneralised sidelobe canceller having a frequency dependent constraintand the objective of uniform directivity over a wide frequency rangewhich is used in the design of the constraint.

The present invention makes use of a substantially logarithmic spiralarray in order to collect raw audio data from as wide a field aspossible. FIG. 2 illustrates a typical logarithmic spiral array.

To construct a microphone array layout of the type shown in FIG. 2, aninner set of microphones 31, 32, 33 are arranged on the vertices of aregular polygon 46 (a triangle is used in the example of FIG. 2) and thespacing between the inner set of microphones is chosen to be less thanone half of the wavelength of the highest frequency at which themicrophone array is expected to operate. Additional microphones 34, 35,36 are added by dilating the basic polygon by a fixed ratio, androtating it so as to maximise the minimum distance between any of thenew microphones and any of the microphones already placed. The ratiobetween the sizes of each successive polygon is chosen to achieve adesired beam width, where a smaller ratio results in a narrower possiblebeam. Microphones are added 37, 38, 39; 40, 41, 42 until the overallsize of the array is sufficient to achieve the desired beam width at thelowest operating frequency, where a larger overall size is required fora narrower beam. This process generally results in an arrangement ofmicrophones with an overall logarithmic spiral form, although theinnermost microphones 31 to 36 may deviate slightly from this patterndepending on the ratio between the sizes of successive polygons and thenumber of vertices of each polygon. This process may produce either aleft-handed or right-handed spiral form 43, 44, 45 and both performequally well.

The advantage of a logarithmic spiral array designed according to thealgorithm described above is that it provides for uniform directivityand sidelobe performance over an extended bandwidth. The design of thelogarithmic spiral array may be further optimised by moving the positionof individual microphone elements, without departing from the overalleffect of the array design.

FIG. 3 illustrates a preferred embodiment of the present invention. Asin FIG. 1, the system comprises a microphone array 1 in a logarithmicspiral arrangement as well as steer delay elements 3 and beam filterelements 4. Optionally, the store 50 may be provided. However, in thisembodiment, the blocking filter comprises a fixed blocking filter 13 andan adaptive blocking filter 12. The output of the fixed blocking filteris passed to the adaptive blocking filter 12 and the outputs of thesteer delay and beam filter elements are summed in summer 10 and thesummer output provides feedback 11 to the adaptive blocking filter 12.

The role of the blocking filter is to constrain the interferencecanceller so that it only cancels out the unwanted, interfering signals,but does not reduce the amplitude of the wanted signal. It does this byeliminating (i.e. nulling) the wanted signal from the microphone signalsto produce a set of interference reference signals. Thus, when theinterference canceller uses the reference signals to cancel theinterference from the output of the fixed beamformer, it does noteliminate the wanted signal at the same time. The blocking filter may,with advantage, be constructed so as to eliminate the wanted signal notonly from a direct acoustic path, but also from known sources ofreflection, such as the ground.

The outputs from the adaptive blocking filter 12 are input to aninterference canceller 14 and the signal from the interference canceller14 is combined with the output of the first summer 10 in a second summer15 to provide an audio feed 16. There is also a feedback 17 of the audiofeed to the interference canceller 14 to improve interferencecancellation. The feedback of the audio signal and the first summationsignal allow the audio output to be fine tuned to remove interferenceeffects from sound sources that are in the direction of interest orreverberant paths from the sound source of interest (i.e. paths arrivingoutside of the main beam).

The present invention uses large numbers of microphone elements in agenerally logarithmic spiral arrangement to provide sufficient coverageto enable a particular sound source to be discriminated, either in realtime or by mean of post processing. The frequency-dependent arrayshading in the beamforming constraint of a generalised sidelobecanceller provides constant beamwidth over a wide frequency range.

By using an adaptive blocking matrix in place of the fixed blockingmatrix used in the basic generalised sidelobe canceller (GSC) of FIG. 1,the susceptibility of the system to leakage of the target signal intothe interference canceller when there are phase errors or variations inthe direction of arrival across the array is reduced, as such leakagewould cause the interference canceller to cancel the target signal. Afixed blocking filter, of the type that has been traditionally used inthe GSC algorithm, followed by the adaptive blocking process as shown inFIG. 3 deals with residual components of the blocked signal that arestill correlated with the target signal thereby providing a furtherenhancement.

The steer delay and beam filter forming a fixed beamformer element ofthe adaptive array allows a real time implementation in broadcastindustry applications. To facilitate rapid steering of the array tofollow the action, a number of beams filters are pre-calculated and theoperator needs only to switch from one to another, with the appropriatedelays switched simultaneously, in order to steer the array toward atarget. Once a target has been identified by listening to the audio feedfrom the fixed beamformer, then the adaptation process can be switchedon to further improve the signal to noise ratio. By recording all theaudio data from all array elements to the store 50, such as hard disk,or at least buffering it for several minutes, the adaptive interferencecancellation can be applied retroactively to any audio data in thechosen direction.

A further advantage of recording all of the array microphone elementsignals is that this gives the broadcaster the ability to remix theaudio presentation during post production with unprecedentedflexibility. Signals that had not been considered significant at thetime of recording can either be enhanced for broadcast, or suppressedaccording to need. A single device according to the present inventioncan be used to generate several distinct simultaneous audio streams,where each stream represents a separate beamform derived from the sameset of array element signals. By this means, all the signals needed fora variety of different multi-speaker audio presentation formats, such asstereo, quadraphonic, 5.1, etc., can be derived with only one microphonearray.

The microphone array of the present invention as applied to thebroadcast industry can be made more straightforward to implement andmore versatile by including in each microphone element 2 its ownanalogue to digital conversion and communication to the centralprocessor unit using a standardised bus. The bus can also provide powerand time synchronisation as illustrated in FIG. 4.

FIG. 4 illustrates a compact array microphone element 18. A microphonecapsule 19 picks up audio signals and amplifies them in a pre-amp 20.The amplified signal is analogue to digital converted in an analogue todigital converter (ADC) 21 and the digital signal 22 is input to a databus interface 23. The ADC clock is provided by a clock generator 24which also provides synchronisation to a bus 26 and the interface 23.The element 18 has its own on-board power supply 25, coupled to the bus26. In this example, the array microphone element comprises a microphonecapsule, digitiser, synchronisation and communications interface andpower supply, all provided in one compact unit, so that each arraymicrophone element connects to a high-bandwidth data bus that suppliespower and synchronisation to the elements, and conveys digitised audioinformation back from each microphone element to the central processormodule. By this means, a relatively small number of connections need beprovided between the microphone array and the processor module.

A particular problem with large scale arrays of this type is incalibration of the array, so it is desirable that the array is set upfor autocalibration. This can be done by providing a sound source thatis separate from the array, and which may be placed in a known positionrelative to the array. One way to achieve accurate positioning is bymounting low-powered collimated laser devices on the array such that thebeams intersect at the desired sound source position. The sound sourcewould be driven to produce a known signal, allowing the relativetransfer functions between each of the array elements to be calculated.

1. An audio recording system, the system comprising an array ofmicrophones and a processor module; wherein the array is a logarithmicspiral array in which the minimum distance between each microphone andall others that lie closer to the centre is maximised; and wherein theprocessor module includes a beam direction control device and a beamwidth control device; whereby an audio target is isolated.
 2. A systemaccording to claim 1, wherein the beam direction control devicecomprises a plurality of switched delay elements.
 3. A system accordingto claim 1, wherein the beam width control device comprises beamfilters.
 4. A system according to claim 1, wherein the processor modulefurther comprises a blocking filter; and an adaptive interferencecanceller.
 5. A system according to claim 1, wherein the processormodule further comprises an adaptive blocking filter for adaptivefiltering of an audio target.
 6. A system according to claim 1, furthercomprising a data store, whereby audio signals from each microphone inthe array are stored for later processing.
 7. A system according toclaim 1, wherein each microphone further comprises an analogue todigital converter and a bus interface, whereby digital data istransferred to the processor module via a bus.